Allow an external audio processing module to be used in WebRTC
[This CL is a rebase of an original CL by solenberg@:
https://codereview.webrtc.org/2948763002/ which in turn was a
rebase of an original CL by peah@:
https://chromium-review.googlesource.com/c/527032/]
Allow an external audio processing module to be used in WebRTC
This CL adds support for optionally using an externally created audio
processing module in a peerconnection. The ownership is shared
between the peerconnection and the external creator of the module.
As part of this the internal ownership of the audio processing module
is moved from VoiceEngine to WebRtcVoiceEngine.
BUG=webrtc:7775
Review-Url: https://codereview.webrtc.org/2961723004
Cr-Commit-Position: refs/heads/master@{#18837}
diff --git a/webrtc/api/peerconnectioninterface.h b/webrtc/api/peerconnectioninterface.h
index ee01212..1b74564 100644
--- a/webrtc/api/peerconnectioninterface.h
+++ b/webrtc/api/peerconnectioninterface.h
@@ -1163,6 +1163,8 @@
// |video_encoder_factory| and |video_decoder_factory| is transferred to the
// returned factory.
//
+// If |audio_mixer| is null, an internal audio mixer will be created and used.
+//
// TODO(deadbeef): Use rtc::scoped_refptr<> and std::unique_ptr<> to make this
// ownership transfer and ref counting more obvious.
//
diff --git a/webrtc/audio/audio_receive_stream_unittest.cc b/webrtc/audio/audio_receive_stream_unittest.cc
index 84efb20..127ea07 100644
--- a/webrtc/audio/audio_receive_stream_unittest.cc
+++ b/webrtc/audio/audio_receive_stream_unittest.cc
@@ -17,6 +17,7 @@
#include "webrtc/audio/conversion.h"
#include "webrtc/call/rtp_stream_receiver_controller.h"
#include "webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h"
+#include "webrtc/modules/audio_processing/include/mock_audio_processing.h"
#include "webrtc/modules/bitrate_controller/include/mock/mock_bitrate_controller.h"
#include "webrtc/modules/pacing/packet_router.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
@@ -74,13 +75,13 @@
RegisterVoiceEngineObserver(_)).WillOnce(Return(0));
EXPECT_CALL(voice_engine_,
DeRegisterVoiceEngineObserver()).WillOnce(Return(0));
- EXPECT_CALL(voice_engine_, audio_processing());
EXPECT_CALL(voice_engine_, audio_device_module());
EXPECT_CALL(voice_engine_, audio_transport());
AudioState::Config config;
config.voice_engine = &voice_engine_;
config.audio_mixer = audio_mixer_;
+ config.audio_processing = new rtc::RefCountedObject<MockAudioProcessing>();
audio_state_ = AudioState::Create(config);
EXPECT_CALL(voice_engine_, ChannelProxyFactory(kChannelId))
diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc
index f89da98..f8ee3ab 100644
--- a/webrtc/audio/audio_send_stream.cc
+++ b/webrtc/audio/audio_send_stream.cc
@@ -279,8 +279,9 @@
stats.audio_level = base->transmit_mixer()->AudioLevelFullRange();
RTC_DCHECK_LE(0, stats.audio_level);
- RTC_DCHECK(base->audio_processing());
- auto audio_processing_stats = base->audio_processing()->GetStatistics();
+ RTC_DCHECK(audio_state_->audio_processing());
+ auto audio_processing_stats =
+ audio_state_->audio_processing()->GetStatistics();
stats.echo_delay_median_ms = audio_processing_stats.delay_median;
stats.echo_delay_std_ms = audio_processing_stats.delay_standard_deviation;
stats.echo_return_loss = audio_processing_stats.echo_return_loss.instant();
diff --git a/webrtc/audio/audio_send_stream_unittest.cc b/webrtc/audio/audio_send_stream_unittest.cc
index e6acf92..efc18d1 100644
--- a/webrtc/audio/audio_send_stream_unittest.cc
+++ b/webrtc/audio/audio_send_stream_unittest.cc
@@ -128,6 +128,7 @@
struct ConfigHelper {
ConfigHelper(bool audio_bwe_enabled, bool expect_set_encoder_call)
: stream_config_(nullptr),
+ audio_processing_(new rtc::RefCountedObject<MockAudioProcessing>()),
simulated_clock_(123456),
send_side_cc_(rtc::MakeUnique<SendSideCongestionController>(
&simulated_clock_,
@@ -144,12 +145,12 @@
EXPECT_CALL(voice_engine_,
DeRegisterVoiceEngineObserver()).WillOnce(Return(0));
EXPECT_CALL(voice_engine_, audio_device_module());
- EXPECT_CALL(voice_engine_, audio_processing());
EXPECT_CALL(voice_engine_, audio_transport());
AudioState::Config config;
config.voice_engine = &voice_engine_;
config.audio_mixer = AudioMixerImpl::Create();
+ config.audio_processing = audio_processing_;
audio_state_ = AudioState::Create(config);
SetupDefaultChannelProxy(audio_bwe_enabled);
@@ -278,8 +279,6 @@
.WillRepeatedly(Return(report_blocks));
EXPECT_CALL(voice_engine_, transmit_mixer())
.WillRepeatedly(Return(&transmit_mixer_));
- EXPECT_CALL(voice_engine_, audio_processing())
- .WillRepeatedly(Return(&audio_processing_));
EXPECT_CALL(transmit_mixer_, AudioLevelFullRange())
.WillRepeatedly(Return(kSpeechInputLevel));
@@ -294,7 +293,7 @@
audio_processing_stats_.delay_median = kEchoDelayMedian;
audio_processing_stats_.delay_standard_deviation = kEchoDelayStdDev;
- EXPECT_CALL(audio_processing_, GetStatistics())
+ EXPECT_CALL(*audio_processing_, GetStatistics())
.WillRepeatedly(Return(audio_processing_stats_));
}
@@ -303,7 +302,7 @@
rtc::scoped_refptr<AudioState> audio_state_;
AudioSendStream::Config stream_config_;
testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr;
- MockAudioProcessing audio_processing_;
+ rtc::scoped_refptr<MockAudioProcessing> audio_processing_;
MockTransmitMixer transmit_mixer_;
AudioProcessing::AudioProcessingStatistics audio_processing_stats_;
SimulatedClock simulated_clock_;
diff --git a/webrtc/audio/audio_state.cc b/webrtc/audio/audio_state.cc
index c15ddd7..961e772 100644
--- a/webrtc/audio/audio_state.cc
+++ b/webrtc/audio/audio_state.cc
@@ -19,11 +19,15 @@
namespace webrtc {
namespace internal {
+// TODO(peah): Remove the conditional in the audio_transport_proxy_ constructor
+// call when upstream dependencies have properly been resolved.
AudioState::AudioState(const AudioState::Config& config)
: config_(config),
voe_base_(config.voice_engine),
audio_transport_proxy_(voe_base_->audio_transport(),
- voe_base_->audio_processing(),
+ config_.audio_processing
+ ? config_.audio_processing.get()
+ : voe_base_->audio_processing(),
config_.audio_mixer) {
process_thread_checker_.DetachFromThread();
RTC_DCHECK(config_.audio_mixer);
diff --git a/webrtc/audio/audio_state.h b/webrtc/audio/audio_state.h
index 29b5a9e..8a98d57 100644
--- a/webrtc/audio/audio_state.h
+++ b/webrtc/audio/audio_state.h
@@ -28,8 +28,14 @@
explicit AudioState(const AudioState::Config& config);
~AudioState() override;
- VoiceEngine* voice_engine();
+ // TODO(peah): Remove the conditional when upstream dependencies have properly
+ // been resolved.
+ AudioProcessing* audio_processing() override {
+ return config_.audio_processing ? config_.audio_processing.get()
+ : voe_base_->audio_processing();
+ }
+ VoiceEngine* voice_engine();
rtc::scoped_refptr<AudioMixer> mixer();
bool typing_noise_detected() const;
diff --git a/webrtc/audio/audio_state_unittest.cc b/webrtc/audio/audio_state_unittest.cc
index 05c5003..ffaacb6 100644
--- a/webrtc/audio/audio_state_unittest.cc
+++ b/webrtc/audio/audio_state_unittest.cc
@@ -12,6 +12,7 @@
#include "webrtc/audio/audio_state.h"
#include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
+#include "webrtc/modules/audio_processing/include/mock_audio_processing.h"
#include "webrtc/test/gtest.h"
#include "webrtc/test/mock_voice_engine.h"
@@ -31,8 +32,6 @@
.WillOnce(testing::Return(0));
EXPECT_CALL(mock_voice_engine, audio_device_module())
.Times(testing::AtLeast(1));
- EXPECT_CALL(mock_voice_engine, audio_processing())
- .Times(testing::AtLeast(1));
EXPECT_CALL(mock_voice_engine, audio_transport())
.WillRepeatedly(testing::Return(&audio_transport));
@@ -49,6 +48,8 @@
audio_state_config.voice_engine = &mock_voice_engine;
audio_state_config.audio_mixer = audio_mixer;
+ audio_state_config.audio_processing =
+ new rtc::RefCountedObject<MockAudioProcessing>();
}
AudioState::Config& config() { return audio_state_config; }
MockVoiceEngine& voice_engine() { return mock_voice_engine; }
diff --git a/webrtc/audio/audio_transport_proxy.cc b/webrtc/audio/audio_transport_proxy.cc
index d6ce939..73b0091 100644
--- a/webrtc/audio/audio_transport_proxy.cc
+++ b/webrtc/audio/audio_transport_proxy.cc
@@ -34,11 +34,13 @@
} // namespace
AudioTransportProxy::AudioTransportProxy(AudioTransport* voe_audio_transport,
- AudioProcessing* apm,
+ AudioProcessing* audio_processing,
AudioMixer* mixer)
- : voe_audio_transport_(voe_audio_transport), apm_(apm), mixer_(mixer) {
+ : voe_audio_transport_(voe_audio_transport),
+ audio_processing_(audio_processing),
+ mixer_(mixer) {
RTC_DCHECK(voe_audio_transport);
- RTC_DCHECK(apm);
+ RTC_DCHECK(audio_processing);
RTC_DCHECK(mixer);
}
@@ -85,7 +87,7 @@
*elapsed_time_ms = mixed_frame_.elapsed_time_ms_;
*ntp_time_ms = mixed_frame_.ntp_time_ms_;
- const auto error = apm_->ProcessReverseStream(&mixed_frame_);
+ const auto error = audio_processing_->ProcessReverseStream(&mixed_frame_);
RTC_DCHECK_EQ(error, AudioProcessing::kNoError);
nSamplesOut = Resample(mixed_frame_, samplesPerSec, &resampler_,
diff --git a/webrtc/audio/audio_transport_proxy.h b/webrtc/audio/audio_transport_proxy.h
index 1d03378..fda9339 100644
--- a/webrtc/audio/audio_transport_proxy.h
+++ b/webrtc/audio/audio_transport_proxy.h
@@ -23,7 +23,7 @@
class AudioTransportProxy : public AudioTransport {
public:
AudioTransportProxy(AudioTransport* voe_audio_transport,
- AudioProcessing* apm,
+ AudioProcessing* audio_processing,
AudioMixer* mixer);
~AudioTransportProxy() override;
@@ -65,7 +65,7 @@
private:
AudioTransport* voe_audio_transport_;
- AudioProcessing* apm_;
+ AudioProcessing* audio_processing_;
rtc::scoped_refptr<AudioMixer> mixer_;
AudioFrame mixed_frame_;
// Converts mixed audio to the audio device output rate.
diff --git a/webrtc/call/audio_state.h b/webrtc/call/audio_state.h
index 2c26a17..32b7ada 100644
--- a/webrtc/call/audio_state.h
+++ b/webrtc/call/audio_state.h
@@ -16,6 +16,7 @@
namespace webrtc {
+class AudioProcessing;
class VoiceEngine;
// WORK IN PROGRESS
@@ -36,8 +37,13 @@
// The audio mixer connected to active receive streams. One per
// AudioState.
rtc::scoped_refptr<AudioMixer> audio_mixer;
+
+ // The audio processing module.
+ rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing;
};
+ virtual AudioProcessing* audio_processing() = 0;
+
// TODO(solenberg): Replace scoped_refptr with shared_ptr once we can use it.
static rtc::scoped_refptr<AudioState> Create(
const AudioState::Config& config);
diff --git a/webrtc/call/call_perf_tests.cc b/webrtc/call/call_perf_tests.cc
index 3b1de73..0c0a0cf 100644
--- a/webrtc/call/call_perf_tests.cc
+++ b/webrtc/call/call_perf_tests.cc
@@ -145,12 +145,15 @@
const uint32_t kAudioRecvSsrc = 5678;
metrics::Reset();
+ rtc::scoped_refptr<AudioProcessing> audio_processing =
+ AudioProcessing::Create();
VoiceEngine* voice_engine = VoiceEngine::Create();
VoEBase* voe_base = VoEBase::GetInterface(voice_engine);
FakeAudioDevice fake_audio_device(
FakeAudioDevice::CreatePulsedNoiseCapturer(256, 48000),
FakeAudioDevice::CreateDiscardRenderer(48000), audio_rtp_speed);
- EXPECT_EQ(0, voe_base->Init(&fake_audio_device, nullptr, decoder_factory_));
+ EXPECT_EQ(0, voe_base->Init(&fake_audio_device, audio_processing.get(),
+ decoder_factory_));
VoEBase::ChannelConfig config;
config.enable_voice_pacing = true;
int send_channel_id = voe_base->CreateChannel(config);
@@ -159,7 +162,9 @@
AudioState::Config send_audio_state_config;
send_audio_state_config.voice_engine = voice_engine;
send_audio_state_config.audio_mixer = AudioMixerImpl::Create();
+ send_audio_state_config.audio_processing = audio_processing;
Call::Config sender_config(event_log_.get());
+
sender_config.audio_state = AudioState::Create(send_audio_state_config);
Call::Config receiver_config(event_log_.get());
receiver_config.audio_state = sender_config.audio_state;
diff --git a/webrtc/call/call_unittest.cc b/webrtc/call/call_unittest.cc
index 8f0a340..5267e7a 100644
--- a/webrtc/call/call_unittest.cc
+++ b/webrtc/call/call_unittest.cc
@@ -37,8 +37,8 @@
webrtc::AudioState::Config audio_state_config;
audio_state_config.voice_engine = &voice_engine_;
audio_state_config.audio_mixer = webrtc::AudioMixerImpl::Create();
+ audio_state_config.audio_processing = webrtc::AudioProcessing::Create();
EXPECT_CALL(voice_engine_, audio_device_module());
- EXPECT_CALL(voice_engine_, audio_processing());
EXPECT_CALL(voice_engine_, audio_transport());
webrtc::Call::Config config(&event_log_);
config.audio_state = webrtc::AudioState::Create(audio_state_config);
@@ -453,11 +453,13 @@
};
ScopedVoiceEngine voice_engine;
- voice_engine.base->Init(&mock_adm);
AudioState::Config audio_state_config;
audio_state_config.voice_engine = voice_engine.voe;
audio_state_config.audio_mixer = mock_mixer;
+ audio_state_config.audio_processing = AudioProcessing::Create();
+ voice_engine.base->Init(&mock_adm, audio_state_config.audio_processing.get());
auto audio_state = AudioState::Create(audio_state_config);
+
RtcEventLogNullImpl event_log;
Call::Config call_config(&event_log);
call_config.audio_state = audio_state;
diff --git a/webrtc/media/base/fakemediaengine.h b/webrtc/media/base/fakemediaengine.h
index 8b0c920..04dfae0 100644
--- a/webrtc/media/base/fakemediaengine.h
+++ b/webrtc/media/base/fakemediaengine.h
@@ -27,6 +27,7 @@
#include "webrtc/media/base/mediaengine.h"
#include "webrtc/media/base/rtputils.h"
#include "webrtc/media/base/streamparams.h"
+#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/p2p/base/sessiondescription.h"
using webrtc::RtpExtension;
@@ -774,7 +775,8 @@
audio_encoder_factory,
const rtc::scoped_refptr<webrtc::AudioDecoderFactory>&
audio_decoder_factory,
- rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer) {
+ rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
+ rtc::scoped_refptr<webrtc::AudioProcessing> apm) {
// Add a fake audio codec. Note that the name must not be "" as there are
// sanity checks against that.
codecs_.push_back(AudioCodec(101, "fake_audio_codec", 0, 0, 1));
@@ -885,6 +887,7 @@
: CompositeMediaEngine<FakeVoiceEngine, FakeVideoEngine>(nullptr,
nullptr,
nullptr,
+ nullptr,
nullptr) {}
virtual ~FakeMediaEngine() {}
diff --git a/webrtc/media/base/mediaengine.h b/webrtc/media/base/mediaengine.h
index ea9ef99..911b971 100644
--- a/webrtc/media/base/mediaengine.h
+++ b/webrtc/media/base/mediaengine.h
@@ -34,6 +34,7 @@
namespace webrtc {
class AudioDeviceModule;
class AudioMixer;
+class AudioProcessing;
class Call;
}
@@ -111,14 +112,19 @@
template<class VOICE, class VIDEO>
class CompositeMediaEngine : public MediaEngineInterface {
public:
- CompositeMediaEngine(webrtc::AudioDeviceModule* adm,
- const rtc::scoped_refptr<webrtc::AudioEncoderFactory>&
- audio_encoder_factory,
- const rtc::scoped_refptr<webrtc::AudioDecoderFactory>&
- audio_decoder_factory,
- rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer)
- : voice_(adm, audio_encoder_factory, audio_decoder_factory, audio_mixer) {
- }
+ CompositeMediaEngine(
+ webrtc::AudioDeviceModule* adm,
+ const rtc::scoped_refptr<webrtc::AudioEncoderFactory>&
+ audio_encoder_factory,
+ const rtc::scoped_refptr<webrtc::AudioDecoderFactory>&
+ audio_decoder_factory,
+ rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
+ rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing)
+ : voice_(adm,
+ audio_encoder_factory,
+ audio_decoder_factory,
+ audio_mixer,
+ audio_processing) {}
virtual ~CompositeMediaEngine() {}
virtual bool Init() {
voice_.Init();
diff --git a/webrtc/media/engine/apm_helpers_unittest.cc b/webrtc/media/engine/apm_helpers_unittest.cc
index 6845d9e..a869903 100644
--- a/webrtc/media/engine/apm_helpers_unittest.cc
+++ b/webrtc/media/engine/apm_helpers_unittest.cc
@@ -33,19 +33,15 @@
// This replicates the conditions from voe_auto_test.
Config config;
config.Set<ExperimentalAgc>(new ExperimentalAgc(false));
+ apm_ = rtc::scoped_refptr<AudioProcessing>(AudioProcessing::Create(config));
EXPECT_EQ(0, voe_wrapper_.base()->Init(
- &mock_audio_device_,
- AudioProcessing::Create(config),
- MockAudioDecoderFactory::CreateEmptyFactory()));
+ &mock_audio_device_, apm_,
+ MockAudioDecoderFactory::CreateEmptyFactory()));
}
- AudioProcessing* apm() {
- return voe_wrapper_.base()->audio_processing();
- }
+ AudioProcessing* apm() { return apm_.get(); }
- const AudioProcessing* apm() const {
- return voe_wrapper_.base()->audio_processing();
- }
+ const AudioProcessing* apm() const { return apm_.get(); }
test::MockAudioDeviceModule* adm() {
return &mock_audio_device_;
@@ -77,6 +73,7 @@
private:
testing::NiceMock<test::MockAudioDeviceModule> mock_audio_device_;
cricket::VoEWrapper voe_wrapper_;
+ rtc::scoped_refptr<AudioProcessing> apm_;
};
} // namespace
diff --git a/webrtc/media/engine/fakewebrtcvoiceengine.h b/webrtc/media/engine/fakewebrtcvoiceengine.h
index 13721ea..f8343f2 100644
--- a/webrtc/media/engine/fakewebrtcvoiceengine.h
+++ b/webrtc/media/engine/fakewebrtcvoiceengine.h
@@ -16,7 +16,6 @@
#include "webrtc/base/checks.h"
#include "webrtc/media/engine/webrtcvoe.h"
-#include "webrtc/modules/audio_processing/include/audio_processing.h"
namespace webrtc {
namespace voe {
@@ -42,10 +41,8 @@
bool neteq_fast_accelerate = false;
};
- explicit FakeWebRtcVoiceEngine(webrtc::AudioProcessing* apm,
- webrtc::voe::TransmitMixer* transmit_mixer)
- : apm_(apm), transmit_mixer_(transmit_mixer) {
- }
+ explicit FakeWebRtcVoiceEngine(webrtc::voe::TransmitMixer* transmit_mixer)
+ : transmit_mixer_(transmit_mixer) {}
~FakeWebRtcVoiceEngine() override {
RTC_CHECK(channels_.empty());
}
@@ -75,9 +72,9 @@
inited_ = false;
return 0;
}
- webrtc::AudioProcessing* audio_processing() override {
- return apm_;
- }
+ // TODO(peah): Remove this when downstream dependencies have properly been
+ // resolved.
+ webrtc::AudioProcessing* audio_processing() override { return nullptr; }
webrtc::AudioDeviceModule* audio_device_module() override {
return nullptr;
}
@@ -131,7 +128,6 @@
int last_channel_ = -1;
std::map<int, Channel*> channels_;
bool fail_create_channel_ = false;
- webrtc::AudioProcessing* apm_ = nullptr;
webrtc::voe::TransmitMixer* transmit_mixer_ = nullptr;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(FakeWebRtcVoiceEngine);
diff --git a/webrtc/media/engine/nullwebrtcvideoengine_unittest.cc b/webrtc/media/engine/nullwebrtcvideoengine_unittest.cc
index 342b5a8..e1a2964 100644
--- a/webrtc/media/engine/nullwebrtcvideoengine_unittest.cc
+++ b/webrtc/media/engine/nullwebrtcvideoengine_unittest.cc
@@ -10,6 +10,7 @@
#include "webrtc/media/engine/nullwebrtcvideoengine.h"
#include "webrtc/media/engine/webrtcvoiceengine.h"
+#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/test/gtest.h"
#include "webrtc/test/mock_audio_decoder_factory.h"
#include "webrtc/test/mock_audio_encoder_factory.h"
@@ -31,7 +32,8 @@
adm,
audio_encoder_factory,
audio_decoder_factory,
- nullptr) {
+ nullptr,
+ webrtc::AudioProcessing::Create()) {
video_.SetExternalDecoderFactory(video_decoder_factory);
video_.SetExternalEncoderFactory(video_encoder_factory);
}
diff --git a/webrtc/media/engine/webrtcmediaengine.cc b/webrtc/media/engine/webrtcmediaengine.cc
index 3c937f2..9e02f58 100644
--- a/webrtc/media/engine/webrtcmediaengine.cc
+++ b/webrtc/media/engine/webrtcmediaengine.cc
@@ -31,26 +31,30 @@
: public CompositeMediaEngine<WebRtcVoiceEngine, NullWebRtcVideoEngine> {
#endif
public:
- WebRtcMediaEngine2(webrtc::AudioDeviceModule* adm,
- const rtc::scoped_refptr<webrtc::AudioEncoderFactory>&
- audio_encoder_factory,
- const rtc::scoped_refptr<webrtc::AudioDecoderFactory>&
- audio_decoder_factory,
- WebRtcVideoEncoderFactory* video_encoder_factory,
- WebRtcVideoDecoderFactory* video_decoder_factory,
- rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer)
+ WebRtcMediaEngine2(
+ webrtc::AudioDeviceModule* adm,
+ const rtc::scoped_refptr<webrtc::AudioEncoderFactory>&
+ audio_encoder_factory,
+ const rtc::scoped_refptr<webrtc::AudioDecoderFactory>&
+ audio_decoder_factory,
+ WebRtcVideoEncoderFactory* video_encoder_factory,
+ WebRtcVideoDecoderFactory* video_decoder_factory,
+ rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
+ rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing)
#ifdef HAVE_WEBRTC_VIDEO
: CompositeMediaEngine<WebRtcVoiceEngine, WebRtcVideoEngine>(
adm,
audio_encoder_factory,
audio_decoder_factory,
- audio_mixer){
+ audio_mixer,
+ audio_processing){
#else
: CompositeMediaEngine<WebRtcVoiceEngine, NullWebRtcVideoEngine>(
adm,
audio_encoder_factory,
audio_decoder_factory,
- audio_mixer) {
+ audio_mixer,
+ audio_processing) {
#endif
video_.SetExternalDecoderFactory(video_decoder_factory);
video_.SetExternalEncoderFactory(video_encoder_factory);
@@ -67,10 +71,11 @@
audio_decoder_factory,
cricket::WebRtcVideoEncoderFactory* video_encoder_factory,
cricket::WebRtcVideoDecoderFactory* video_decoder_factory,
- rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer) {
+ rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
+ rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing) {
return new cricket::WebRtcMediaEngine2(
adm, audio_encoder_factory, audio_decoder_factory, video_encoder_factory,
- video_decoder_factory, audio_mixer);
+ video_decoder_factory, audio_mixer, audio_processing);
}
void DestroyWebRtcMediaEngine(cricket::MediaEngineInterface* media_engine) {
@@ -89,7 +94,7 @@
return CreateWebRtcMediaEngine(
adm, webrtc::CreateBuiltinAudioEncoderFactory(),
webrtc::CreateBuiltinAudioDecoderFactory(), video_encoder_factory,
- video_decoder_factory, nullptr);
+ video_decoder_factory, nullptr, webrtc::AudioProcessing::Create());
}
MediaEngineInterface* WebRtcMediaEngineFactory::Create(
@@ -100,7 +105,8 @@
WebRtcVideoDecoderFactory* video_decoder_factory) {
return CreateWebRtcMediaEngine(
adm, webrtc::CreateBuiltinAudioEncoderFactory(), audio_decoder_factory,
- video_encoder_factory, video_decoder_factory, nullptr);
+ video_encoder_factory, video_decoder_factory, nullptr,
+ webrtc::AudioProcessing::Create());
}
// Used by PeerConnectionFactory to create a media engine passed into
@@ -111,10 +117,12 @@
audio_decoder_factory,
WebRtcVideoEncoderFactory* video_encoder_factory,
WebRtcVideoDecoderFactory* video_decoder_factory,
- rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer) {
+ rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
+ rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing) {
return CreateWebRtcMediaEngine(
adm, webrtc::CreateBuiltinAudioEncoderFactory(), audio_decoder_factory,
- video_encoder_factory, video_decoder_factory, audio_mixer);
+ video_encoder_factory, video_decoder_factory, audio_mixer,
+ audio_processing);
}
MediaEngineInterface* WebRtcMediaEngineFactory::Create(
@@ -125,9 +133,9 @@
audio_decoder_factory,
WebRtcVideoEncoderFactory* video_encoder_factory,
WebRtcVideoDecoderFactory* video_decoder_factory) {
- return CreateWebRtcMediaEngine(adm, audio_encoder_factory,
- audio_decoder_factory, video_encoder_factory,
- video_decoder_factory, nullptr);
+ return CreateWebRtcMediaEngine(
+ adm, audio_encoder_factory, audio_decoder_factory, video_encoder_factory,
+ video_decoder_factory, nullptr, webrtc::AudioProcessing::Create());
}
MediaEngineInterface* WebRtcMediaEngineFactory::Create(
@@ -138,10 +146,11 @@
audio_decoder_factory,
WebRtcVideoEncoderFactory* video_encoder_factory,
WebRtcVideoDecoderFactory* video_decoder_factory,
- rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer) {
- return CreateWebRtcMediaEngine(adm, audio_encoder_factory,
- audio_decoder_factory, video_encoder_factory,
- video_decoder_factory, audio_mixer);
+ rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
+ rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing) {
+ return CreateWebRtcMediaEngine(
+ adm, audio_encoder_factory, audio_decoder_factory, video_encoder_factory,
+ video_decoder_factory, audio_mixer, audio_processing);
}
namespace {
diff --git a/webrtc/media/engine/webrtcmediaengine.h b/webrtc/media/engine/webrtcmediaengine.h
index db955d0..edbda66 100644
--- a/webrtc/media/engine/webrtcmediaengine.h
+++ b/webrtc/media/engine/webrtcmediaengine.h
@@ -22,6 +22,7 @@
class AudioDecoderFactory;
class AudioDeviceModule;
class AudioMixer;
+class AudioProcessing;
}
namespace cricket {
class WebRtcVideoDecoderFactory;
@@ -59,7 +60,8 @@
audio_decoder_factory,
WebRtcVideoEncoderFactory* video_encoder_factory,
WebRtcVideoDecoderFactory* video_decoder_factory,
- rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer);
+ rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
+ rtc::scoped_refptr<webrtc::AudioProcessing> apm);
static MediaEngineInterface* Create(
webrtc::AudioDeviceModule* adm,
@@ -78,7 +80,8 @@
audio_decoder_factory,
WebRtcVideoEncoderFactory* video_encoder_factory,
WebRtcVideoDecoderFactory* video_decoder_factory,
- rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer);
+ rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
+ rtc::scoped_refptr<webrtc::AudioProcessing> apm);
};
// Verify that extension IDs are within 1-byte extension range and are not
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index b39be5b..1f97949 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -163,7 +163,8 @@
webrtc::AudioState::Config MakeAudioStateConfig(
VoEWrapper* voe_wrapper,
- rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer) {
+ rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
+ rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing) {
webrtc::AudioState::Config config;
config.voice_engine = voe_wrapper->engine();
if (audio_mixer) {
@@ -171,6 +172,7 @@
} else {
config.audio_mixer = webrtc::AudioMixerImpl::Create();
}
+ config.audio_processing = audio_processing;
return config;
}
@@ -214,11 +216,13 @@
webrtc::AudioDeviceModule* adm,
const rtc::scoped_refptr<webrtc::AudioEncoderFactory>& encoder_factory,
const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
- rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer)
+ rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
+ rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing)
: WebRtcVoiceEngine(adm,
encoder_factory,
decoder_factory,
audio_mixer,
+ audio_processing,
nullptr) {}
WebRtcVoiceEngine::WebRtcVoiceEngine(
@@ -226,11 +230,13 @@
const rtc::scoped_refptr<webrtc::AudioEncoderFactory>& encoder_factory,
const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
+ rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing,
VoEWrapper* voe_wrapper)
: adm_(adm),
encoder_factory_(encoder_factory),
decoder_factory_(decoder_factory),
audio_mixer_(audio_mixer),
+ apm_(audio_processing),
voe_wrapper_(voe_wrapper) {
// This may be called from any thread, so detach thread checkers.
worker_thread_checker_.DetachFromThread();
@@ -238,6 +244,7 @@
LOG(LS_INFO) << "WebRtcVoiceEngine::WebRtcVoiceEngine";
RTC_DCHECK(decoder_factory);
RTC_DCHECK(encoder_factory);
+ RTC_DCHECK(audio_processing);
// The rest of our initialization will happen in Init.
}
@@ -284,8 +291,8 @@
webrtc::Trace::SetTraceCallback(this);
webrtc::Trace::set_level_filter(kElevatedTraceFilter);
LOG(LS_INFO) << webrtc::VoiceEngine::GetVersionString();
- RTC_CHECK_EQ(0, voe_wrapper_->base()->Init(adm_.get(), nullptr,
- decoder_factory_));
+ RTC_CHECK_EQ(0,
+ voe_wrapper_->base()->Init(adm_.get(), apm(), decoder_factory_));
webrtc::Trace::set_level_filter(kDefaultTraceFilter);
// No ADM supplied? Get the default one from VoE.
@@ -294,15 +301,12 @@
}
RTC_DCHECK(adm_);
- apm_ = voe_wrapper_->base()->audio_processing();
- RTC_DCHECK(apm_);
-
transmit_mixer_ = voe_wrapper_->base()->transmit_mixer();
RTC_DCHECK(transmit_mixer_);
// Save the default AGC configuration settings. This must happen before
// calling ApplyOptions or the default will be overwritten.
- default_agc_config_ = webrtc::apm_helpers::GetAgcConfig(apm_);
+ default_agc_config_ = webrtc::apm_helpers::GetAgcConfig(apm());
// Set default engine options.
{
@@ -336,8 +340,8 @@
// May be null for VoE injected for testing.
if (voe()->engine()) {
- audio_state_ =
- webrtc::AudioState::Create(MakeAudioStateConfig(voe(), audio_mixer_));
+ audio_state_ = webrtc::AudioState::Create(
+ MakeAudioStateConfig(voe(), audio_mixer_, apm_));
}
initialized_ = true;
@@ -507,7 +511,7 @@
<< default_agc_config_.targetLeveldBOv << "dB to -"
<< config.targetLeveldBOv << "dB";
}
- webrtc::apm_helpers::SetAgcConfig(apm_, config);
+ webrtc::apm_helpers::SetAgcConfig(apm(), config);
}
if (options.intelligibility_enhancer) {
@@ -750,8 +754,7 @@
webrtc::AudioProcessing* WebRtcVoiceEngine::apm() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
- RTC_DCHECK(apm_);
- return apm_;
+ return apm_.get();
}
webrtc::voe::TransmitMixer* WebRtcVoiceEngine::transmit_mixer() {
diff --git a/webrtc/media/engine/webrtcvoiceengine.h b/webrtc/media/engine/webrtcvoiceengine.h
index cad2006..d154d0a 100644
--- a/webrtc/media/engine/webrtcvoiceengine.h
+++ b/webrtc/media/engine/webrtcvoiceengine.h
@@ -57,13 +57,15 @@
webrtc::AudioDeviceModule* adm,
const rtc::scoped_refptr<webrtc::AudioEncoderFactory>& encoder_factory,
const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
- rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer);
+ rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
+ rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing);
// Dependency injection for testing.
WebRtcVoiceEngine(
webrtc::AudioDeviceModule* adm,
const rtc::scoped_refptr<webrtc::AudioEncoderFactory>& encoder_factory,
const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
+ rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing,
VoEWrapper* voe_wrapper);
~WebRtcVoiceEngine() override;
@@ -133,7 +135,7 @@
rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory_;
rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer_;
// Reference to the APM, owned by VoE.
- webrtc::AudioProcessing* apm_ = nullptr;
+ rtc::scoped_refptr<webrtc::AudioProcessing> apm_;
// Reference to the TransmitMixer, owned by VoE.
webrtc::voe::TransmitMixer* transmit_mixer_ = nullptr;
// The primary instance of WebRtc VoiceEngine.
diff --git a/webrtc/media/engine/webrtcvoiceengine_unittest.cc b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
index dbee6d3..85158d8 100644
--- a/webrtc/media/engine/webrtcvoiceengine_unittest.cc
+++ b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
@@ -15,6 +15,7 @@
#include "webrtc/base/arraysize.h"
#include "webrtc/base/byteorder.h"
#include "webrtc/base/safe_conversions.h"
+#include "webrtc/base/scoped_ref_ptr.h"
#include "webrtc/call/call.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
#include "webrtc/media/base/fakemediaengine.h"
@@ -121,19 +122,21 @@
TEST(WebRtcVoiceEngineTestStubLibrary, StartupShutdown) {
StrictMock<webrtc::test::MockAudioDeviceModule> adm;
AdmSetupExpectations(&adm);
- StrictMock<webrtc::test::MockAudioProcessing> apm;
- EXPECT_CALL(apm, ApplyConfig(testing::_));
- EXPECT_CALL(apm, SetExtraOptions(testing::_));
- EXPECT_CALL(apm, Initialize()).WillOnce(Return(0));
- EXPECT_CALL(apm, DetachAecDump());
+ rtc::scoped_refptr<StrictMock<webrtc::test::MockAudioProcessing>> apm =
+ new rtc::RefCountedObject<
+ StrictMock<webrtc::test::MockAudioProcessing>>();
+ EXPECT_CALL(*apm, ApplyConfig(testing::_));
+ EXPECT_CALL(*apm, SetExtraOptions(testing::_));
+ EXPECT_CALL(*apm, Initialize()).WillOnce(Return(0));
+ EXPECT_CALL(*apm, DetachAecDump());
StrictMock<MockTransmitMixer> transmit_mixer;
EXPECT_CALL(transmit_mixer, EnableStereoChannelSwapping(false));
- cricket::FakeWebRtcVoiceEngine voe(&apm, &transmit_mixer);
+ cricket::FakeWebRtcVoiceEngine voe(&transmit_mixer);
EXPECT_FALSE(voe.IsInited());
{
cricket::WebRtcVoiceEngine engine(
&adm, webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
- webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr,
+ webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm,
new FakeVoEWrapper(&voe));
engine.Init();
EXPECT_TRUE(voe.IsInited());
@@ -155,17 +158,22 @@
WebRtcVoiceEngineTestFake() : WebRtcVoiceEngineTestFake("") {}
explicit WebRtcVoiceEngineTestFake(const char* field_trials)
- : apm_gc_(*apm_.gain_control()), apm_ec_(*apm_.echo_cancellation()),
- apm_ns_(*apm_.noise_suppression()), apm_vd_(*apm_.voice_detection()),
- call_(webrtc::Call::Config(&event_log_)), voe_(&apm_, &transmit_mixer_),
+ : apm_(new rtc::RefCountedObject<
+ StrictMock<webrtc::test::MockAudioProcessing>>()),
+ apm_gc_(*apm_->gain_control()),
+ apm_ec_(*apm_->echo_cancellation()),
+ apm_ns_(*apm_->noise_suppression()),
+ apm_vd_(*apm_->voice_detection()),
+ call_(webrtc::Call::Config(&event_log_)),
+ voe_(&transmit_mixer_),
override_field_trials_(field_trials) {
// AudioDeviceModule.
AdmSetupExpectations(&adm_);
// AudioProcessing.
- EXPECT_CALL(apm_, ApplyConfig(testing::_));
- EXPECT_CALL(apm_, SetExtraOptions(testing::_));
- EXPECT_CALL(apm_, Initialize()).WillOnce(Return(0));
- EXPECT_CALL(apm_, DetachAecDump());
+ EXPECT_CALL(*apm_, ApplyConfig(testing::_));
+ EXPECT_CALL(*apm_, SetExtraOptions(testing::_));
+ EXPECT_CALL(*apm_, Initialize()).WillOnce(Return(0));
+ EXPECT_CALL(*apm_, DetachAecDump());
// Default Options.
EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0));
EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0));
@@ -186,7 +194,7 @@
auto encoder_factory = webrtc::CreateBuiltinAudioEncoderFactory();
auto decoder_factory = webrtc::CreateBuiltinAudioDecoderFactory();
engine_.reset(new cricket::WebRtcVoiceEngine(&adm_, encoder_factory,
- decoder_factory, nullptr,
+ decoder_factory, nullptr, apm_,
new FakeVoEWrapper(&voe_)));
engine_->Init();
send_parameters_.codecs.push_back(kPcmuCodec);
@@ -196,8 +204,8 @@
}
bool SetupChannel() {
- EXPECT_CALL(apm_, ApplyConfig(testing::_));
- EXPECT_CALL(apm_, SetExtraOptions(testing::_));
+ EXPECT_CALL(*apm_, ApplyConfig(testing::_));
+ EXPECT_CALL(*apm_, SetExtraOptions(testing::_));
channel_ = engine_->CreateChannel(&call_, cricket::MediaConfig(),
cricket::AudioOptions());
return (channel_ != nullptr);
@@ -217,7 +225,7 @@
if (!channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcX))) {
return false;
}
- EXPECT_CALL(apm_, set_output_will_be_muted(false));
+ EXPECT_CALL(*apm_, set_output_will_be_muted(false));
return channel_->SetAudioSend(kSsrcX, true, nullptr, &fake_source_);
}
@@ -270,26 +278,26 @@
EXPECT_CALL(adm_, RecordingIsInitialized()).WillOnce(Return(false));
EXPECT_CALL(adm_, Recording()).WillOnce(Return(false));
EXPECT_CALL(adm_, InitRecording()).WillOnce(Return(0));
- EXPECT_CALL(apm_, ApplyConfig(testing::_));
- EXPECT_CALL(apm_, SetExtraOptions(testing::_));
+ EXPECT_CALL(*apm_, ApplyConfig(testing::_));
+ EXPECT_CALL(*apm_, SetExtraOptions(testing::_));
}
channel_->SetSend(enable);
}
void SetSendParameters(const cricket::AudioSendParameters& params) {
- EXPECT_CALL(apm_, ApplyConfig(testing::_));
- EXPECT_CALL(apm_, SetExtraOptions(testing::_));
+ EXPECT_CALL(*apm_, ApplyConfig(testing::_));
+ EXPECT_CALL(*apm_, SetExtraOptions(testing::_));
ASSERT_TRUE(channel_);
EXPECT_TRUE(channel_->SetSendParameters(params));
}
void SetAudioSend(uint32_t ssrc, bool enable, cricket::AudioSource* source,
const cricket::AudioOptions* options = nullptr) {
- EXPECT_CALL(apm_, set_output_will_be_muted(!enable));
+ EXPECT_CALL(*apm_, set_output_will_be_muted(!enable));
ASSERT_TRUE(channel_);
if (enable && options) {
- EXPECT_CALL(apm_, ApplyConfig(testing::_));
- EXPECT_CALL(apm_, SetExtraOptions(testing::_));
+ EXPECT_CALL(*apm_, ApplyConfig(testing::_));
+ EXPECT_CALL(*apm_, SetExtraOptions(testing::_));
}
EXPECT_TRUE(channel_->SetAudioSend(ssrc, enable, options, source));
}
@@ -658,7 +666,7 @@
protected:
StrictMock<webrtc::test::MockAudioDeviceModule> adm_;
- StrictMock<webrtc::test::MockAudioProcessing> apm_;
+ rtc::scoped_refptr<StrictMock<webrtc::test::MockAudioProcessing>> apm_;
webrtc::test::MockGainControl& apm_gc_;
webrtc::test::MockEchoCancellation& apm_ec_;
webrtc::test::MockNoiseSuppression& apm_ns_;
@@ -2794,8 +2802,8 @@
RecordingIsInitialized()).Times(2).WillRepeatedly(Return(false));
EXPECT_CALL(adm_, Recording()).Times(2).WillRepeatedly(Return(false));
EXPECT_CALL(adm_, InitRecording()).Times(2).WillRepeatedly(Return(0));
- EXPECT_CALL(apm_, ApplyConfig(testing::_)).Times(10);
- EXPECT_CALL(apm_, SetExtraOptions(testing::_)).Times(10);
+ EXPECT_CALL(*apm_, ApplyConfig(testing::_)).Times(10);
+ EXPECT_CALL(*apm_, SetExtraOptions(testing::_)).Times(10);
std::unique_ptr<cricket::WebRtcVoiceMediaChannel> channel1(
static_cast<cricket::WebRtcVoiceMediaChannel*>(engine_->CreateChannel(
@@ -2905,8 +2913,8 @@
cricket::MediaConfig config;
std::unique_ptr<cricket::VoiceMediaChannel> channel;
- EXPECT_CALL(apm_, ApplyConfig(testing::_)).Times(3);
- EXPECT_CALL(apm_, SetExtraOptions(testing::_)).Times(3);
+ EXPECT_CALL(*apm_, ApplyConfig(testing::_)).Times(3);
+ EXPECT_CALL(*apm_, SetExtraOptions(testing::_)).Times(3);
channel.reset(
engine_->CreateChannel(&call_, config, cricket::AudioOptions()));
@@ -3262,9 +3270,11 @@
TEST(WebRtcVoiceEngineTest, StartupShutdown) {
// If the VoiceEngine wants to gather available codecs early, that's fine but
// we never want it to create a decoder at this stage.
+ rtc::scoped_refptr<webrtc::AudioProcessing> apm =
+ webrtc::AudioProcessing::Create();
cricket::WebRtcVoiceEngine engine(
nullptr, webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
- webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr);
+ webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm);
engine.Init();
webrtc::RtcEventLogNullImpl event_log;
std::unique_ptr<webrtc::Call> call(
@@ -3284,9 +3294,11 @@
// we could enter a tight loop since the mock would return 0.
EXPECT_CALL(adm, TimeUntilNextProcess()).WillRepeatedly(Return(100));
{
+ rtc::scoped_refptr<webrtc::AudioProcessing> apm =
+ webrtc::AudioProcessing::Create();
cricket::WebRtcVoiceEngine engine(
&adm, webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
- webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr);
+ webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm);
engine.Init();
webrtc::RtcEventLogNullImpl event_log;
std::unique_ptr<webrtc::Call> call(
@@ -3302,9 +3314,11 @@
TEST(WebRtcVoiceEngineTest, HasCorrectPayloadTypeMapping) {
// TODO(ossu): Why are the payload types of codecs with non-static payload
// type assignments checked here? It shouldn't really matter.
+ rtc::scoped_refptr<webrtc::AudioProcessing> apm =
+ webrtc::AudioProcessing::Create();
cricket::WebRtcVoiceEngine engine(
nullptr, webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
- webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr);
+ webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm);
engine.Init();
for (const cricket::AudioCodec& codec : engine.send_codecs()) {
auto is_codec = [&codec](const char* name, int clockrate = 0) {
@@ -3344,9 +3358,11 @@
// Tests that VoE supports at least 32 channels
TEST(WebRtcVoiceEngineTest, Has32Channels) {
+ rtc::scoped_refptr<webrtc::AudioProcessing> apm =
+ webrtc::AudioProcessing::Create();
cricket::WebRtcVoiceEngine engine(
nullptr, webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
- webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr);
+ webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm);
engine.Init();
webrtc::RtcEventLogNullImpl event_log;
std::unique_ptr<webrtc::Call> call(
@@ -3379,9 +3395,11 @@
// what we sent in - though it's probably reasonable to expect so, if
// SetRecvParameters returns true.
// I think it will become clear once audio decoder injection is completed.
+ rtc::scoped_refptr<webrtc::AudioProcessing> apm =
+ webrtc::AudioProcessing::Create();
cricket::WebRtcVoiceEngine engine(
nullptr, webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
- webrtc::CreateBuiltinAudioDecoderFactory(), nullptr);
+ webrtc::CreateBuiltinAudioDecoderFactory(), nullptr, apm);
engine.Init();
webrtc::RtcEventLogNullImpl event_log;
std::unique_ptr<webrtc::Call> call(
@@ -3418,8 +3436,10 @@
EXPECT_CALL(*mock_decoder_factory.get(), GetSupportedDecoders())
.WillOnce(Return(specs));
+ rtc::scoped_refptr<webrtc::AudioProcessing> apm =
+ webrtc::AudioProcessing::Create();
cricket::WebRtcVoiceEngine engine(nullptr, unused_encoder_factory,
- mock_decoder_factory, nullptr);
+ mock_decoder_factory, nullptr, apm);
engine.Init();
auto codecs = engine.recv_codecs();
EXPECT_EQ(11, codecs.size());
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc
index f3c5557..e151c43 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
@@ -321,7 +321,8 @@
AudioProcessing* AudioProcessing::Create(const webrtc::Config& config,
NonlinearBeamformer* beamformer) {
- AudioProcessingImpl* apm = new AudioProcessingImpl(config, beamformer);
+ AudioProcessingImpl* apm =
+ new rtc::RefCountedObject<AudioProcessingImpl>(config, beamformer);
if (apm->Initialize() != kNoError) {
delete apm;
apm = nullptr;
diff --git a/webrtc/modules/audio_processing/audio_processing_impl_unittest.cc b/webrtc/modules/audio_processing/audio_processing_impl_unittest.cc
index ca6b7f6..2ee1d51 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl_unittest.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl_unittest.cc
@@ -20,6 +20,7 @@
using ::testing::Return;
namespace webrtc {
+namespace {
class MockInitialize : public AudioProcessingImpl {
public:
@@ -30,8 +31,13 @@
int RealInitializeLocked() NO_THREAD_SAFETY_ANALYSIS {
return AudioProcessingImpl::InitializeLocked();
}
+
+ MOCK_CONST_METHOD0(AddRef, int());
+ MOCK_CONST_METHOD0(Release, int());
};
+} // namespace
+
TEST(AudioProcessingImplTest, AudioParameterChangeTriggersInit) {
webrtc::Config config;
MockInitialize mock(config);
diff --git a/webrtc/modules/audio_processing/audio_processing_unittest.cc b/webrtc/modules/audio_processing/audio_processing_unittest.cc
index 11ce917..59fbb91 100644
--- a/webrtc/modules/audio_processing/audio_processing_unittest.cc
+++ b/webrtc/modules/audio_processing/audio_processing_unittest.cc
@@ -7,7 +7,6 @@
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
-
#include <math.h>
#include <stdio.h>
@@ -2805,7 +2804,7 @@
// the config, and that the default initial level is maintained after the
// config has been applied.
std::unique_ptr<AudioProcessingImpl> apm(
- new AudioProcessingImpl(webrtc::Config()));
+ new rtc::RefCountedObject<AudioProcessingImpl>(webrtc::Config()));
AudioProcessing::Config config;
EXPECT_FALSE(apm->config_.level_controller.enabled);
// TODO(peah): Add test for the existence of the level controller object once
@@ -2835,7 +2834,7 @@
// Verify that the initial level can be specified and is retained after the
// config has been applied.
std::unique_ptr<AudioProcessingImpl> apm(
- new AudioProcessingImpl(webrtc::Config()));
+ new rtc::RefCountedObject<AudioProcessingImpl>(webrtc::Config()));
AudioProcessing::Config config;
config.level_controller.initial_peak_level_dbfs = -50.f;
apm->ApplyConfig(config);
@@ -2857,7 +2856,7 @@
// Verify that the config is properly reset when the specified initial peak
// level is too low.
std::unique_ptr<AudioProcessingImpl> apm(
- new AudioProcessingImpl(webrtc::Config()));
+ new rtc::RefCountedObject<AudioProcessingImpl>(webrtc::Config()));
AudioProcessing::Config config;
config.level_controller.enabled = true;
config.level_controller.initial_peak_level_dbfs = -101.f;
@@ -2875,7 +2874,7 @@
// Verify that the config is properly reset when the specified initial peak
// level is too high.
- apm.reset(new AudioProcessingImpl(webrtc::Config()));
+ apm.reset(new rtc::RefCountedObject<AudioProcessingImpl>(webrtc::Config()));
config = AudioProcessing::Config();
config.level_controller.enabled = true;
config.level_controller.initial_peak_level_dbfs = 1.f;
diff --git a/webrtc/modules/audio_processing/include/audio_processing.h b/webrtc/modules/audio_processing/include/audio_processing.h
index 6a5a81d..05be5fe 100644
--- a/webrtc/modules/audio_processing/include/audio_processing.h
+++ b/webrtc/modules/audio_processing/include/audio_processing.h
@@ -21,6 +21,7 @@
#include "webrtc/base/arraysize.h"
#include "webrtc/base/platform_file.h"
+#include "webrtc/base/refcount.h"
#include "webrtc/modules/audio_processing/beamformer/array_util.h"
#include "webrtc/modules/audio_processing/include/config.h"
#include "webrtc/typedefs.h"
@@ -233,7 +234,7 @@
// // Close the application...
// delete apm;
//
-class AudioProcessing {
+class AudioProcessing : public rtc::RefCountInterface {
public:
// The struct below constitutes the new parameter scheme for the audio
// processing. It is being introduced gradually and until it is fully
@@ -300,7 +301,7 @@
// Only for testing.
static AudioProcessing* Create(const webrtc::Config& config,
NonlinearBeamformer* beamformer);
- virtual ~AudioProcessing() {}
+ ~AudioProcessing() override {}
// Initializes internal states, while retaining all user settings. This
// should be called before beginning to process a new audio stream. However,
diff --git a/webrtc/ortc/BUILD.gn b/webrtc/ortc/BUILD.gn
index fd8631c..b6a2cc9 100644
--- a/webrtc/ortc/BUILD.gn
+++ b/webrtc/ortc/BUILD.gn
@@ -41,6 +41,7 @@
"../logging:rtc_event_log_api",
"../media:rtc_media",
"../media:rtc_media_base",
+ "../modules/audio_processing:audio_processing",
"../p2p:rtc_p2p",
"../pc:libjingle_peerconnection",
"../pc:rtc_pc",
diff --git a/webrtc/ortc/DEPS b/webrtc/ortc/DEPS
index de152dd..4e18244 100644
--- a/webrtc/ortc/DEPS
+++ b/webrtc/ortc/DEPS
@@ -5,6 +5,7 @@
"+webrtc/logging/rtc_event_log",
"+webrtc/media",
"+webrtc/modules/audio_coding",
+ "+webrtc/modules/audio_processing",
"+webrtc/p2p",
"+webrtc/pc",
diff --git a/webrtc/ortc/ortcfactory.cc b/webrtc/ortc/ortcfactory.cc
index bf34f13..612e55d 100644
--- a/webrtc/ortc/ortcfactory.cc
+++ b/webrtc/ortc/ortcfactory.cc
@@ -27,6 +27,7 @@
#include "webrtc/base/logging.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
#include "webrtc/media/base/mediaconstants.h"
+#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/ortc/ortcrtpreceiveradapter.h"
#include "webrtc/ortc/ortcrtpsenderadapter.h"
#include "webrtc/ortc/rtpparametersconversion.h"
@@ -544,9 +545,9 @@
// Note that |adm_| may be null, in which case the platform-specific default
// AudioDeviceModule will be used.
return std::unique_ptr<cricket::MediaEngineInterface>(
- cricket::WebRtcMediaEngineFactory::Create(adm_, audio_encoder_factory_,
- audio_decoder_factory_, nullptr,
- nullptr, nullptr));
+ cricket::WebRtcMediaEngineFactory::Create(
+ adm_, audio_encoder_factory_, audio_decoder_factory_, nullptr,
+ nullptr, nullptr, webrtc::AudioProcessing::Create()));
}
} // namespace webrtc
diff --git a/webrtc/pc/BUILD.gn b/webrtc/pc/BUILD.gn
index 60d76ed..5190a93 100644
--- a/webrtc/pc/BUILD.gn
+++ b/webrtc/pc/BUILD.gn
@@ -203,6 +203,7 @@
"../logging:rtc_event_log_api",
"../media:rtc_audio_video",
"../modules/audio_device:audio_device",
+ "../modules/audio_processing:audio_processing",
]
configs += [ ":libjingle_peerconnection_warnings_config" ]
diff --git a/webrtc/pc/DEPS b/webrtc/pc/DEPS
index 4468e6b..d77d279 100644
--- a/webrtc/pc/DEPS
+++ b/webrtc/pc/DEPS
@@ -8,6 +8,7 @@
"+webrtc/logging/rtc_event_log",
"+webrtc/media",
"+webrtc/modules/audio_device",
+ "+webrtc/modules/audio_processing",
"+webrtc/modules/rtp_rtcp",
"+webrtc/modules/video_coding",
"+webrtc/modules/video_render",
diff --git a/webrtc/pc/createpeerconnectionfactory.cc b/webrtc/pc/createpeerconnectionfactory.cc
index 9370c0b..c29ef52 100644
--- a/webrtc/pc/createpeerconnectionfactory.cc
+++ b/webrtc/pc/createpeerconnectionfactory.cc
@@ -17,6 +17,7 @@
#include "webrtc/call/callfactoryinterface.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log_factory_interface.h"
#include "webrtc/media/engine/webrtcmediaengine.h"
+#include "webrtc/modules/audio_processing/include/audio_processing.h"
namespace webrtc {
@@ -53,7 +54,8 @@
std::unique_ptr<cricket::MediaEngineInterface> media_engine(
cricket::WebRtcMediaEngineFactory::Create(
default_adm, audio_encoder_factory, audio_decoder_factory,
- video_encoder_factory, video_decoder_factory, audio_mixer));
+ video_encoder_factory, video_decoder_factory, audio_mixer,
+ AudioProcessing::Create()));
std::unique_ptr<CallFactoryInterface> call_factory = CreateCallFactory();
diff --git a/webrtc/pc/peerconnectioninterface_unittest.cc b/webrtc/pc/peerconnectioninterface_unittest.cc
index 1229f40..c36460f 100644
--- a/webrtc/pc/peerconnectioninterface_unittest.cc
+++ b/webrtc/pc/peerconnectioninterface_unittest.cc
@@ -30,6 +30,7 @@
#include "webrtc/media/base/fakevideocapturer.h"
#include "webrtc/media/engine/webrtcmediaengine.h"
#include "webrtc/media/sctp/sctptransportinternal.h"
+#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/p2p/base/fakeportallocator.h"
#include "webrtc/pc/audiotrack.h"
#include "webrtc/pc/mediasession.h"
@@ -650,7 +651,7 @@
auto media_engine = std::unique_ptr<cricket::MediaEngineInterface>(
cricket::WebRtcMediaEngineFactory::Create(
nullptr, audio_encoder_factory, audio_decoder_factory, nullptr,
- nullptr, nullptr));
+ nullptr, nullptr, webrtc::AudioProcessing::Create()));
std::unique_ptr<webrtc::CallFactoryInterface> call_factory =
webrtc::CreateCallFactory();
diff --git a/webrtc/sdk/android/BUILD.gn b/webrtc/sdk/android/BUILD.gn
index a2c0dd4..689d6cf 100644
--- a/webrtc/sdk/android/BUILD.gn
+++ b/webrtc/sdk/android/BUILD.gn
@@ -183,6 +183,7 @@
"//webrtc/call:call_interfaces",
"//webrtc/logging:rtc_event_log_api",
"//webrtc/media:rtc_audio_video",
+ "//webrtc/modules/audio_processing:audio_processing",
]
if (is_clang) {
@@ -325,8 +326,8 @@
deps = [
":libjingle_peerconnection_java",
":libjingle_peerconnection_metrics_default_java",
- "//webrtc/rtc_base:base_java",
"//webrtc/modules/audio_device:audio_device_java",
+ "//webrtc/rtc_base:base_java",
]
}
@@ -414,8 +415,8 @@
]
deps = [
- "//webrtc/rtc_base:base_java",
"//webrtc/modules/audio_device:audio_device_java",
+ "//webrtc/rtc_base:base_java",
]
}
diff --git a/webrtc/sdk/android/src/jni/DEPS b/webrtc/sdk/android/src/jni/DEPS
index 803232c..ed8a8f4 100644
--- a/webrtc/sdk/android/src/jni/DEPS
+++ b/webrtc/sdk/android/src/jni/DEPS
@@ -7,6 +7,7 @@
"+webrtc/logging/rtc_event_log/rtc_event_log_factory_interface.h",
"+webrtc/media/base",
"+webrtc/media/engine",
+ "+webrtc/modules/audio_processing/include/audio_processing.h",
"+webrtc/modules/utility/include/jvm_android.h",
"+webrtc/modules/video_coding/include/video_codec_interface.h",
"+webrtc/modules/video_coding/utility/vp8_header_parser.h",
diff --git a/webrtc/sdk/android/src/jni/media_jni.cc b/webrtc/sdk/android/src/jni/media_jni.cc
index 0bc2737..fdcdc0f 100644
--- a/webrtc/sdk/android/src/jni/media_jni.cc
+++ b/webrtc/sdk/android/src/jni/media_jni.cc
@@ -12,6 +12,7 @@
#include "webrtc/call/callfactoryinterface.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log_factory_interface.h"
#include "webrtc/media/engine/webrtcmediaengine.h"
+#include "webrtc/modules/audio_processing/include/audio_processing.h"
namespace webrtc_jni {
@@ -34,7 +35,7 @@
rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer) {
return cricket::WebRtcMediaEngineFactory::Create(
adm, audio_encoder_factory, audio_decoder_factory, video_encoder_factory,
- video_decoder_factory, audio_mixer);
+ video_decoder_factory, audio_mixer, webrtc::AudioProcessing::Create());
}
} // namespace webrtc_jni
diff --git a/webrtc/test/call_test.cc b/webrtc/test/call_test.cc
index 6e665cf..638e704 100644
--- a/webrtc/test/call_test.cc
+++ b/webrtc/test/call_test.cc
@@ -56,10 +56,13 @@
CreateFakeAudioDevices(test->CreateCapturer(), test->CreateRenderer());
test->OnFakeAudioDevicesCreated(fake_send_audio_device_.get(),
fake_recv_audio_device_.get());
+ apm_send_ = AudioProcessing::Create();
+ apm_recv_ = AudioProcessing::Create();
CreateVoiceEngines();
AudioState::Config audio_state_config;
audio_state_config.voice_engine = voe_send_.voice_engine;
audio_state_config.audio_mixer = AudioMixerImpl::Create();
+ audio_state_config.audio_processing = apm_send_;
send_config.audio_state = AudioState::Create(audio_state_config);
}
CreateSenderCall(send_config);
@@ -69,6 +72,7 @@
AudioState::Config audio_state_config;
audio_state_config.voice_engine = voe_recv_.voice_engine;
audio_state_config.audio_mixer = AudioMixerImpl::Create();
+ audio_state_config.audio_processing = apm_recv_;
recv_config.audio_state = AudioState::Create(audio_state_config);
}
CreateReceiverCall(recv_config);
@@ -378,8 +382,8 @@
void CallTest::CreateVoiceEngines() {
voe_send_.voice_engine = VoiceEngine::Create();
voe_send_.base = VoEBase::GetInterface(voe_send_.voice_engine);
- EXPECT_EQ(0, voe_send_.base->Init(fake_send_audio_device_.get(), nullptr,
- decoder_factory_));
+ EXPECT_EQ(0, voe_send_.base->Init(fake_send_audio_device_.get(),
+ apm_send_.get(), decoder_factory_));
VoEBase::ChannelConfig config;
config.enable_voice_pacing = true;
voe_send_.channel_id = voe_send_.base->CreateChannel(config);
@@ -387,8 +391,8 @@
voe_recv_.voice_engine = VoiceEngine::Create();
voe_recv_.base = VoEBase::GetInterface(voe_recv_.voice_engine);
- EXPECT_EQ(0, voe_recv_.base->Init(fake_recv_audio_device_.get(), nullptr,
- decoder_factory_));
+ EXPECT_EQ(0, voe_recv_.base->Init(fake_recv_audio_device_.get(),
+ apm_recv_.get(), decoder_factory_));
voe_recv_.channel_id = voe_recv_.base->CreateChannel();
EXPECT_GE(voe_recv_.channel_id, 0);
}
diff --git a/webrtc/test/call_test.h b/webrtc/test/call_test.h
index 39df343..e7b75d6 100644
--- a/webrtc/test/call_test.h
+++ b/webrtc/test/call_test.h
@@ -146,6 +146,8 @@
VoiceEngineState voe_send_;
VoiceEngineState voe_recv_;
+ rtc::scoped_refptr<AudioProcessing> apm_send_;
+ rtc::scoped_refptr<AudioProcessing> apm_recv_;
// The audio devices must outlive the voice engines.
std::unique_ptr<test::FakeAudioDevice> fake_send_audio_device_;
diff --git a/webrtc/test/mock_voice_engine.h b/webrtc/test/mock_voice_engine.h
index 7655d3b..e8eb5a2 100644
--- a/webrtc/test/mock_voice_engine.h
+++ b/webrtc/test/mock_voice_engine.h
@@ -15,7 +15,6 @@
#include "webrtc/modules/audio_device/include/mock_audio_device.h"
#include "webrtc/modules/audio_device/include/mock_audio_transport.h"
-#include "webrtc/modules/audio_processing/include/mock_audio_processing.h"
#include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
#include "webrtc/test/gmock.h"
#include "webrtc/test/mock_voe_channel_proxy.h"
@@ -68,8 +67,6 @@
.WillByDefault(testing::Return(1000));
ON_CALL(*this, audio_device_module())
.WillByDefault(testing::Return(&mock_audio_device_));
- ON_CALL(*this, audio_processing())
- .WillByDefault(testing::Return(&mock_audio_processing_));
ON_CALL(*this, audio_transport())
.WillByDefault(testing::Return(&mock_audio_transport_));
}
@@ -102,9 +99,8 @@
MOCK_METHOD3(
Init,
int(AudioDeviceModule* external_adm,
- AudioProcessing* audioproc,
+ AudioProcessing* external_apm,
const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory));
- MOCK_METHOD0(audio_processing, AudioProcessing*());
MOCK_METHOD0(audio_device_module, AudioDeviceModule*());
MOCK_METHOD0(transmit_mixer, voe::TransmitMixer*());
MOCK_METHOD0(Terminate, int());
@@ -257,7 +253,6 @@
std::map<int, std::unique_ptr<MockRtpRtcp>> mock_rtp_rtcps_;
MockAudioDeviceModule mock_audio_device_;
- MockAudioProcessing mock_audio_processing_;
MockAudioTransport mock_audio_transport_;
};
} // namespace test
diff --git a/webrtc/video/video_quality_test.cc b/webrtc/video/video_quality_test.cc
index 37b4786..3f0465d 100644
--- a/webrtc/video/video_quality_test.cc
+++ b/webrtc/video/video_quality_test.cc
@@ -79,12 +79,13 @@
int receive_channel_id;
};
-void CreateVoiceEngine(VoiceEngineState* voe,
- rtc::scoped_refptr<webrtc::AudioDecoderFactory>
- decoder_factory) {
+void CreateVoiceEngine(
+ VoiceEngineState* voe,
+ webrtc::AudioProcessing* apm,
+ rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory) {
voe->voice_engine = webrtc::VoiceEngine::Create();
voe->base = webrtc::VoEBase::GetInterface(voe->voice_engine);
- EXPECT_EQ(0, voe->base->Init(nullptr, nullptr, decoder_factory));
+ EXPECT_EQ(0, voe->base->Init(nullptr, apm, decoder_factory));
webrtc::VoEBase::ChannelConfig config;
config.enable_voice_pacing = true;
voe->send_channel_id = voe->base->CreateChannel(config);
@@ -1849,11 +1850,15 @@
call_config.bitrate_config = params_.call.call_bitrate_config;
::VoiceEngineState voe;
+ rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing(
+ webrtc::AudioProcessing::Create());
+
if (params_.audio.enabled) {
- CreateVoiceEngine(&voe, decoder_factory_);
+ CreateVoiceEngine(&voe, audio_processing.get(), decoder_factory_);
AudioState::Config audio_state_config;
audio_state_config.voice_engine = voe.voice_engine;
audio_state_config.audio_mixer = AudioMixerImpl::Create();
+ audio_state_config.audio_processing = audio_processing;
call_config.audio_state = AudioState::Create(audio_state_config);
}
diff --git a/webrtc/voice_engine/include/voe_base.h b/webrtc/voice_engine/include/voe_base.h
index dec07d6..05fe455 100644
--- a/webrtc/voice_engine/include/voe_base.h
+++ b/webrtc/voice_engine/include/voe_base.h
@@ -124,18 +124,19 @@
// modules:
// - The Audio Device Module (ADM) which implements all the audio layer
// functionality in a separate (reference counted) module.
- // - The AudioProcessing module handles capture-side processing. VoiceEngine
- // takes ownership of this object.
+ // - The AudioProcessing module handles capture-side processing.
// - An AudioDecoderFactory - used to create audio decoders.
- // If NULL is passed for any of these, VoiceEngine will create its own.
- // Returns -1 in case of an error, 0 otherwise.
+ // If NULL is passed for either of ADM or AudioDecoderFactory, VoiceEngine
+ // will create its own. Returns -1 in case of an error, 0 otherwise.
// TODO(ajm): Remove default NULLs.
virtual int Init(AudioDeviceModule* external_adm = NULL,
- AudioProcessing* audioproc = NULL,
+ AudioProcessing* external_apm = nullptr,
const rtc::scoped_refptr<AudioDecoderFactory>&
decoder_factory = nullptr) = 0;
- // Returns NULL before Init() is called.
+ // Returns null before Init() is called.
+ // TODO(peah): Remove this when downstream dependencies have properly been
+ // resolved.
virtual AudioProcessing* audio_processing() = 0;
// This method is WIP - DO NOT USE!
diff --git a/webrtc/voice_engine/shared_data.cc b/webrtc/voice_engine/shared_data.cc
index e77f52a..464374e 100644
--- a/webrtc/voice_engine/shared_data.cc
+++ b/webrtc/voice_engine/shared_data.cc
@@ -61,7 +61,6 @@
}
void SharedData::set_audio_processing(AudioProcessing* audioproc) {
- audioproc_.reset(audioproc);
_transmitMixerPtr->SetAudioProcessingModule(audioproc);
_outputMixerPtr->SetAudioProcessingModule(audioproc);
}
diff --git a/webrtc/voice_engine/shared_data.h b/webrtc/voice_engine/shared_data.h
index 1a91040..951a851 100644
--- a/webrtc/voice_engine/shared_data.h
+++ b/webrtc/voice_engine/shared_data.h
@@ -43,7 +43,6 @@
AudioDeviceModule* audio_device() { return _audioDevicePtr.get(); }
void set_audio_device(
const rtc::scoped_refptr<AudioDeviceModule>& audio_device);
- AudioProcessing* audio_processing() { return audioproc_.get(); }
void set_audio_processing(AudioProcessing* audio_processing);
TransmitMixer* transmit_mixer() { return _transmitMixerPtr; }
OutputMixer* output_mixer() { return _outputMixerPtr; }
@@ -69,7 +68,6 @@
rtc::scoped_refptr<AudioDeviceModule> _audioDevicePtr;
OutputMixer* _outputMixerPtr;
TransmitMixer* _transmitMixerPtr;
- std::unique_ptr<AudioProcessing> audioproc_;
std::unique_ptr<ProcessThread> _moduleProcessThreadPtr;
// |encoder_queue| is defined last to ensure all pending tasks are cancelled
// and deleted before any other members.
diff --git a/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc b/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc
index 6fbf5f1..f7510c7 100644
--- a/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc
+++ b/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc
@@ -55,6 +55,9 @@
local_network_ = webrtc::VoENetwork::GetInterface(local_voe_);
local_rtp_rtcp_ = webrtc::VoERTP_RTCP::GetInterface(local_voe_);
+ local_apm_ = webrtc::AudioProcessing::Create();
+ local_base_->Init(nullptr, local_apm_.get(), nullptr);
+
// In principle, we can use one VoiceEngine to achieve the same goal. Well, in
// here, we use two engines to make it more like reality.
remote_voe_ = webrtc::VoiceEngine::Create();
@@ -64,7 +67,9 @@
remote_rtp_rtcp_ = webrtc::VoERTP_RTCP::GetInterface(remote_voe_);
remote_file_ = webrtc::VoEFile::GetInterface(remote_voe_);
- EXPECT_EQ(0, local_base_->Init());
+ remote_apm_ = webrtc::AudioProcessing::Create();
+ remote_base_->Init(nullptr, remote_apm_.get(), nullptr);
+
local_sender_ = local_base_->CreateChannel();
static_cast<webrtc::VoiceEngineImpl*>(local_voe_)
->GetChannelProxy(local_sender_)
@@ -74,10 +79,8 @@
EXPECT_EQ(0, local_rtp_rtcp_->
SetSendAudioLevelIndicationStatus(local_sender_, true,
kAudioLevelHeaderId));
-
EXPECT_EQ(0, local_base_->StartSend(local_sender_));
- EXPECT_EQ(0, remote_base_->Init());
reflector_ = remote_base_->CreateChannel();
static_cast<webrtc::VoiceEngineImpl*>(remote_voe_)
->GetChannelProxy(reflector_)
diff --git a/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h b/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h
index 37919e1..19fb15f 100644
--- a/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h
+++ b/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h
@@ -147,6 +147,7 @@
webrtc::VoEBase* local_base_;
webrtc::VoERTP_RTCP* local_rtp_rtcp_;
webrtc::VoENetwork* local_network_;
+ rtc::scoped_refptr<webrtc::AudioProcessing> local_apm_;
webrtc::VoiceEngine* remote_voe_;
webrtc::VoEBase* remote_base_;
@@ -154,7 +155,7 @@
webrtc::VoERTP_RTCP* remote_rtp_rtcp_;
webrtc::VoENetwork* remote_network_;
webrtc::VoEFile* remote_file_;
-
+ rtc::scoped_refptr<webrtc::AudioProcessing> remote_apm_;
LoudestFilter loudest_filter_;
const std::unique_ptr<webrtc::RtpHeaderParser> rtp_header_parser_;
diff --git a/webrtc/voice_engine/test/auto_test/standard/codec_test.cc b/webrtc/voice_engine/test/auto_test/standard/codec_test.cc
index 2b5d12d..27c3b0e 100644
--- a/webrtc/voice_engine/test/auto_test/standard/codec_test.cc
+++ b/webrtc/voice_engine/test/auto_test/standard/codec_test.cc
@@ -19,6 +19,8 @@
protected:
void SetUp() {
memset(&codec_instance_, 0, sizeof(codec_instance_));
+ apm_ = webrtc::AudioProcessing::Create();
+ voe_base_->Init(nullptr, apm_.get(), nullptr);
}
void SetArbitrarySendCodec() {
@@ -27,6 +29,7 @@
EXPECT_EQ(0, voe_codec_->SetSendCodec(channel_, codec_instance_));
}
+ rtc::scoped_refptr<webrtc::AudioProcessing> apm_;
webrtc::CodecInst codec_instance_;
};
diff --git a/webrtc/voice_engine/voe_base_impl.cc b/webrtc/voice_engine/voe_base_impl.cc
index 1ddf53c..196c6a5 100644
--- a/webrtc/voice_engine/voe_base_impl.cc
+++ b/webrtc/voice_engine/voe_base_impl.cc
@@ -225,8 +225,10 @@
int VoEBaseImpl::Init(
AudioDeviceModule* external_adm,
- AudioProcessing* audioproc,
+ AudioProcessing* external_apm,
const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) {
+ // TODO(peah): Add a DCHECK for external_apm when downstream dependencies
+ // have properly been resolved.
rtc::CritScope cs(shared_->crit_sec());
WebRtcSpl_Init();
if (shared_->statistics().Initialized()) {
@@ -337,33 +339,43 @@
"Init() failed to set mono/stereo recording mode");
}
- if (!audioproc) {
- audioproc = AudioProcessing::Create();
- if (!audioproc) {
+ // TODO(peah): Remove this when upstream dependencies have properly been
+ // resolved.
+ AudioProcessing* apm = nullptr;
+ if (!external_apm) {
+ audio_processing_ = AudioProcessing::Create();
+ if (!audio_processing_) {
+ // This can only happen if there are problems allocating the dynamic
+ // memory in the Create() call.
LOG(LS_ERROR) << "Failed to create AudioProcessing.";
shared_->SetLastError(VE_NO_MEMORY);
return -1;
}
+ apm = audio_processing_.get();
+ } else {
+ apm = external_apm;
}
- shared_->set_audio_processing(audioproc);
+
+ shared_->set_audio_processing(apm);
// Set the error state for any failures in this block.
shared_->SetLastError(VE_APM_ERROR);
// Configure AudioProcessing components.
- if (audioproc->high_pass_filter()->Enable(true) != 0) {
+ // TODO(peah): Move this initialization to webrtcvoiceengine.cc.
+ if (apm->high_pass_filter()->Enable(true) != 0) {
LOG_F(LS_ERROR) << "Failed to enable high pass filter.";
return -1;
}
- if (audioproc->echo_cancellation()->enable_drift_compensation(false) != 0) {
+ if (apm->echo_cancellation()->enable_drift_compensation(false) != 0) {
LOG_F(LS_ERROR) << "Failed to disable drift compensation.";
return -1;
}
- if (audioproc->noise_suppression()->set_level(kDefaultNsMode) != 0) {
+ if (apm->noise_suppression()->set_level(kDefaultNsMode) != 0) {
LOG_F(LS_ERROR) << "Failed to set noise suppression level: "
<< kDefaultNsMode;
return -1;
}
- GainControl* agc = audioproc->gain_control();
+ GainControl* agc = apm->gain_control();
if (agc->set_analog_level_limits(kMinVolumeLevel, kMaxVolumeLevel) != 0) {
LOG_F(LS_ERROR) << "Failed to set analog level limits with minimum: "
<< kMinVolumeLevel << " and maximum: " << kMaxVolumeLevel;
@@ -687,9 +699,7 @@
shared_->set_audio_device(nullptr);
}
- if (shared_->audio_processing()) {
- shared_->set_audio_processing(nullptr);
- }
+ shared_->set_audio_processing(nullptr);
return shared_->statistics().SetUnInitialized();
}
diff --git a/webrtc/voice_engine/voe_base_impl.h b/webrtc/voice_engine/voe_base_impl.h
index b9f3282..6d7a086 100644
--- a/webrtc/voice_engine/voe_base_impl.h
+++ b/webrtc/voice_engine/voe_base_impl.h
@@ -28,12 +28,15 @@
int RegisterVoiceEngineObserver(VoiceEngineObserver& observer) override;
int DeRegisterVoiceEngineObserver() override;
- int Init(AudioDeviceModule* external_adm = nullptr,
- AudioProcessing* audioproc = nullptr,
- const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory =
- nullptr) override;
+ int Init(
+ AudioDeviceModule* external_adm,
+ AudioProcessing* external_apm,
+ const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) override;
AudioProcessing* audio_processing() override {
- return shared_->audio_processing();
+ // TODO(peah): Remove this when downstream dependencies have properly been
+ // resolved.
+ RTC_DCHECK(audio_processing_);
+ return audio_processing_.get();
}
AudioDeviceModule* audio_device_module() override {
return shared_->audio_device();
@@ -121,6 +124,10 @@
rtc::CriticalSection callbackCritSect_;
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
+ // TODO(peah): Remove this when upstream dependencies have properly been
+ // resolved.
+ rtc::scoped_refptr<AudioProcessing> audio_processing_;
+
AudioFrame audioFrame_;
voe::SharedData* shared_;
};
diff --git a/webrtc/voice_engine/voe_base_unittest.cc b/webrtc/voice_engine/voe_base_unittest.cc
index a300f3b..55f9634 100644
--- a/webrtc/voice_engine/voe_base_unittest.cc
+++ b/webrtc/voice_engine/voe_base_unittest.cc
@@ -21,17 +21,8 @@
class VoEBaseTest : public VoiceEngineFixture {};
-TEST_F(VoEBaseTest, InitWithExternalAudioDeviceAndAudioProcessing) {
- AudioProcessing* audioproc = AudioProcessing::Create();
- EXPECT_EQ(0, base_->Init(&adm_, audioproc));
- EXPECT_EQ(audioproc, base_->audio_processing());
- EXPECT_EQ(0, base_->LastError());
-}
-
TEST_F(VoEBaseTest, InitWithExternalAudioDevice) {
- EXPECT_EQ(nullptr, base_->audio_processing());
- EXPECT_EQ(0, base_->Init(&adm_, nullptr));
- EXPECT_NE(nullptr, base_->audio_processing());
+ EXPECT_EQ(0, base_->Init(&adm_, apm_.get()));
EXPECT_EQ(0, base_->LastError());
}
@@ -41,15 +32,14 @@
}
TEST_F(VoEBaseTest, CreateChannelAfterInit) {
- EXPECT_EQ(0, base_->Init(&adm_, nullptr));
+ EXPECT_EQ(0, base_->Init(&adm_, apm_.get(), nullptr));
int channelID = base_->CreateChannel();
EXPECT_NE(channelID, -1);
EXPECT_EQ(0, base_->DeleteChannel(channelID));
}
TEST_F(VoEBaseTest, AssociateSendChannel) {
- AudioProcessing* audioproc = AudioProcessing::Create();
- EXPECT_EQ(0, base_->Init(&adm_, audioproc));
+ EXPECT_EQ(0, base_->Init(&adm_, apm_.get()));
const int channel_1 = base_->CreateChannel();
diff --git a/webrtc/voice_engine/voe_codec_unittest.cc b/webrtc/voice_engine/voe_codec_unittest.cc
index af04a61..7d27785 100644
--- a/webrtc/voice_engine/voe_codec_unittest.cc
+++ b/webrtc/voice_engine/voe_codec_unittest.cc
@@ -13,6 +13,7 @@
#include "webrtc/voice_engine/include/voe_codec.h"
#include "webrtc/modules/audio_device/include/fake_audio_device.h"
+#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/test/gtest.h"
#include "webrtc/voice_engine/include/voe_base.h"
#include "webrtc/voice_engine/voice_engine_defines.h"
@@ -79,8 +80,9 @@
VoEBase* base(VoEBase::GetInterface(voe));
VoECodec* voe_codec(VoECodec::GetInterface(voe));
std::unique_ptr<FakeAudioDeviceModule> adm(new FakeAudioDeviceModule);
+ std::unique_ptr<AudioProcessing> apm(AudioProcessing::Create());
- base->Init(adm.get());
+ base->Init(adm.get(), apm.get());
CodecInst codec = {111, "opus", 48000, 960, 1, 32000};
diff --git a/webrtc/voice_engine/voe_network_unittest.cc b/webrtc/voice_engine/voe_network_unittest.cc
index ef6e4fb..d3d1f80 100644
--- a/webrtc/voice_engine/voe_network_unittest.cc
+++ b/webrtc/voice_engine/voe_network_unittest.cc
@@ -32,7 +32,7 @@
class VoENetworkTest : public VoiceEngineFixture {
protected:
int CreateChannelAndRegisterExternalTransport() {
- EXPECT_EQ(0, base_->Init(&adm_, nullptr));
+ EXPECT_EQ(0, base_->Init(&adm_, apm_.get(), nullptr));
int channelID = base_->CreateChannel();
EXPECT_NE(channelID, -1);
EXPECT_EQ(0, network_->RegisterExternalTransport(channelID, transport_));
@@ -47,19 +47,19 @@
TEST_F(VoENetworkTest,
RegisterExternalTransportOnNonExistingChannelShouldFail) {
- EXPECT_EQ(0, base_->Init(&adm_, nullptr));
+ EXPECT_EQ(0, base_->Init(&adm_, apm_.get(), nullptr));
EXPECT_NE(
0, network_->RegisterExternalTransport(kNonExistingChannel, transport_));
}
TEST_F(VoENetworkTest,
DeRegisterExternalTransportOnNonExistingChannelShouldFail) {
- EXPECT_EQ(0, base_->Init(&adm_, nullptr));
+ EXPECT_EQ(0, base_->Init(&adm_, apm_.get(), nullptr));
EXPECT_NE(0, network_->DeRegisterExternalTransport(kNonExistingChannel));
}
TEST_F(VoENetworkTest, DeRegisterExternalTransportBeforeRegister) {
- EXPECT_EQ(0, base_->Init(&adm_, nullptr));
+ EXPECT_EQ(0, base_->Init(&adm_, apm_.get(), nullptr));
int channelID = base_->CreateChannel();
EXPECT_NE(channelID, -1);
EXPECT_EQ(0, network_->DeRegisterExternalTransport(channelID));
@@ -72,13 +72,13 @@
}
TEST_F(VoENetworkTest, ReceivedRTPPacketOnNonExistingChannelShouldFail) {
- EXPECT_EQ(0, base_->Init(&adm_, nullptr));
+ EXPECT_EQ(0, base_->Init(&adm_, apm_.get(), nullptr));
EXPECT_EQ(-1, network_->ReceivedRTPPacket(kNonExistingChannel, kPacket,
sizeof(kPacket)));
}
TEST_F(VoENetworkTest, ReceivedRTPPacketOnChannelWithoutTransportShouldFail) {
- EXPECT_EQ(0, base_->Init(&adm_, nullptr));
+ EXPECT_EQ(0, base_->Init(&adm_, apm_.get(), nullptr));
int channelID = base_->CreateChannel();
EXPECT_NE(channelID, -1);
EXPECT_EQ(-1,
@@ -105,13 +105,13 @@
}
TEST_F(VoENetworkTest, ReceivedRTCPPacketOnNonExistingChannelShouldFail) {
- EXPECT_EQ(0, base_->Init(&adm_, nullptr));
+ EXPECT_EQ(0, base_->Init(&adm_, apm_.get(), nullptr));
EXPECT_EQ(-1, network_->ReceivedRTCPPacket(kNonExistingChannel, kPacket,
sizeof(kPacket)));
}
TEST_F(VoENetworkTest, ReceivedRTCPPacketOnChannelWithoutTransportShouldFail) {
- EXPECT_EQ(0, base_->Init(&adm_, nullptr));
+ EXPECT_EQ(0, base_->Init(&adm_, apm_.get(), nullptr));
int channelID = base_->CreateChannel();
EXPECT_NE(channelID, -1);
EXPECT_EQ(-1,
diff --git a/webrtc/voice_engine/voice_engine_fixture.cc b/webrtc/voice_engine/voice_engine_fixture.cc
index faac8dd..3cb613d 100644
--- a/webrtc/voice_engine/voice_engine_fixture.cc
+++ b/webrtc/voice_engine/voice_engine_fixture.cc
@@ -9,6 +9,7 @@
*/
#include "webrtc/voice_engine/voice_engine_fixture.h"
+#include "webrtc/modules/audio_processing/include/mock_audio_processing.h"
namespace webrtc {
@@ -19,6 +20,7 @@
EXPECT_NE(nullptr, base_);
EXPECT_NE(nullptr, network_);
EXPECT_EQ(0, base_->RegisterVoiceEngineObserver(observer_));
+ apm_ = new rtc::RefCountedObject<test::MockAudioProcessing>();
}
VoiceEngineFixture::~VoiceEngineFixture() {
diff --git a/webrtc/voice_engine/voice_engine_fixture.h b/webrtc/voice_engine/voice_engine_fixture.h
index 4f4eaa8..9f2f316 100644
--- a/webrtc/voice_engine/voice_engine_fixture.h
+++ b/webrtc/voice_engine/voice_engine_fixture.h
@@ -9,6 +9,7 @@
*/
#include "webrtc/modules/audio_device/include/fake_audio_device.h"
+#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/test/gtest.h"
#include "webrtc/test/mock_transport.h"
#include "webrtc/voice_engine/include/voe_base.h"
@@ -28,6 +29,7 @@
MockVoEObserver observer_;
FakeAudioDeviceModule adm_;
MockTransport transport_;
+ rtc::scoped_refptr<AudioProcessing> apm_;
};
} // namespace webrtc
